1. Field
This disclosure relates to voice over packet network calling, more particularly to a method to allow users to select less expensive voice over packet network calling at a later time or a more expensive public switched transmission network immediately.
2. Background
Packet voice networks allow owners of data networks to use those data networks to place long-distance calls and avoid the typical long-distance tariffs. The data network may have access to the Internet or a proprietary network through an access provider, for which a flat fee is paid. Typically there is no fee associated with the amount of traffic generated by terminals on the network. Large networks commonly have enough excess capacity that information other than data can be transmitted with no further charges being levied.
One type of such data is voice data. In a traditional phone call, the caller and recipient speak to each other through a public switched transmission network (PSTN). In this network, the caller identifies the recipient by dialing a number and a direct link is established between the two parties. Voice signals travel this link with almost no delay, typically providing good quality signals on both ends.
In a packet voice network, the voice signal is coded into digital form and possibly compressed. Compression allows more data to be packed into fewer bits, resulting in fewer packets being created. After the voice data is coded, it is packetized into packets for the given network type and transmitted as if it were data. At the receiving end, the packets are reassembled and converted back to voice signals. Each packet can travel a different path, and may actually arrive at the destination out of order. The system orders them and decodes them to produce the voice signal of the caller.
One major concern in packet networks is quality of service (QoS). The direct links of a PSTN service offer high quality transmission with little or no delay. Packet voice networks need to provide similar quality to achieve acceptance. However, several factors can contribute to degradation of voice transmissions on packet networks.
The process of coding and decoding the packets can cause delay of two types. The first type is related to the transmission delay of the packets. This may disrupt the rhythm of the conversation to the point when the caller and recipient are talking out of turn because one or the other did not receive the voice signals in enough time to avoid talking over the other. The second type of delay, referred to as jitter, is a disruption of the pattern of speech. Speech has a distinct rhythm in which pauses can have as much meaning as words. Jitter disrupts this rhythm and can further degrade the quality of service.
In either case, part of the problem lies in the network delays in getting packets to their destination in a timely manner. The main obstacle to timely delivery is network congestion. If the network devices, such as routers and servers, have queues of packets waiting to be sent, and the connections are running at maximum bandwidth, delays will occur in packet networks. This will lead to delays in transmitting and receiving voice packets, degrading the QoS of the packet voice network. Another problem is that some packets are lost during their transit of the network. These lost packets further degrade the QoS of packet voice networks.
However, using a packet voice network has the advantage of being far less expensive than PSTN. This makes the use of packet voice desirable. One solution that has developed in several companies is to try the packet voice network first. If the parameters of the packet voice network are not within those needed to achieve a voice call with high QoS, the system typically switches to a PSTN fallback link, if one is configured. While this will allow a large percentage of the attempted voice over packet network calls to be completed, a significant number of them will be rejected due to network congestion.
Network congestion can clear up in a relatively short time, depending upon the speed of the network devices and the bandwidth available. The ability to make the call over the less expensive voice network may be missed by a few seconds because of the fallback process. It would be more efficient if a technique were available that allowed the packet network to be tested repeatedly and the call completed using the packet network when bandwidth becomes available and can be guaranteed.